PJSIP and UDP and cellphone networks

slvrsi · October 1, 2020, 2:38pm

Having difficulties getting UDP to work over cell networks

So I have asterisk 16 installed using PJSIP but it doesnt work over cell networks, it works over regular networks so it must be something that the cellnetwork is doing to the packet? but; i’m not sure.
Cell network is FIDO in Canada if anyone knows anything about it.

Basicly can’t send register packets over UDP; but works over TCP. If I can get registry, I can’t get voice because of course that uses UDP. This is only over the cell network.

slvrsi · October 1, 2020, 2:41pm

pjsip_chan logger on.

<— Received SIP request (635 bytes) from UDP:FIDO.IP:20372 —>
REGISTER sip:ASTERISKEXTHOSTNAME SIP/2.0
Via: SIP/2.0/UDP NoClueWhatIpThisIs-fidogateway?-25.109.120.46:5080;branch=z9hG4bK.OmsF6NghP;rport
From: sip:username@ASTERISKEXTHOSTNAME;tag=lcHCQ7S7U
To: sip:username@ASTERISKEXTHOSTNAME
CSeq: 20 REGISTER
Call-ID: bcjR7YcPFZ
Max-Forwards: 70u
Supported: replaces, outbound, gruu
Accept: application/sdp
Accept: text/plain
Accept: application/vnd.gsma.rcs-ft-http+xml
Contact: sip:username@NoClueWhatIpThisIs-fidogateway?-25.109.120.46:5080;transport=udp;+sip.instance=“urn:uuid:883f5ef2-690e-0058-9ebe-c04afeb8fa0b”;+org.linphone.specs=“groupchat,lime”
Expires: 3600
User-Agent: LinphoneAndroid/4.2 (Pixel 2) LinphoneSDK/4.3 (tags/4.3^0)

<— Transmitting SIP response (475 bytes) to UDP:FIDO.IP:20372 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP NoClueWhatIpThisIs-fidogateway?-25.109.120.46:5080;rport=20372;received=FIDO.IP;branch=z9hG4bK.OmsF6NghP
Call-ID: bcjR7YcPFZ
From: sip:username@ASTERISKEXTHOSTNAME;tag=lcHCQ7S7U
To: sip:username@ASTERISKEXTHOSTNAME;tag=z9hG4bK.OmsF6NghP
CSeq: 20 REGISTER
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1601559616/a7bac9b2dd972764d199712a3be03c03”,opaque=“1fc8f0854bd2f521”,algorithm=md5,qop=“auth”
Server: Asterisk PBX 16.3.0
Content-Length: 0

This above just repeats… when on WIFI only - it does show

<— Received SIP request (918 bytes) from UDP:phoneIP:5080 —>
REGISTER

and continues to register…

jcolp · October 1, 2020, 2:42pm

What client are you using? What is the PJSIP configuration?

slvrsi · October 1, 2020, 2:46pm

Wow that was quick. I was trying to post the logger stuff but 2 link limits so I was modifying it.

I’ve tried a bunch with the same result but using linphone right now on android.
User-Agent: LinphoneAndroid/4.2 (Pixel 2) LinphoneSDK/4.3 (tags/4.3^0)

jcolp · October 1, 2020, 2:48pm

You should be able to post things now.

slvrsi · October 1, 2020, 2:51pm

thanks!. posted. above.

pjsip_wizard:
username
endpoint/context = *****
inbound_auth/password=*****
inbound_auth/username=****
endpoint/force_rport = yes
endpoint/rewrite_contact = yes
endpoint/rtp_symmetric=yes
endpoint/direct_media = no
aor/minimum_expiration = 600

user_defaults
type=wizard
;transport = ipv4
accepts_registrations = yes
sends_registrations = no
accepts_auth = yes
sends_auth = no
endpoint/allow_subscribe = yes
endpoint/allow = !all,ulaw,gsm,g722
endpoint/direct_media = no
endpoint/force_rport = yes
endpoint/disable_direct_media_on_nat = yes
endpoint/direct_media_method = invite
endpoint/ice_support = no
endpoint/moh_suggest = default
endpoint/send_rpid = yes
endpoint/rewrite_contact = yes
endpoint/send_pai = yes
endpoint/allow_transfer = yes
endpoint/trust_id_inbound = yes
endpoint/trust_id_outbound = yes
endpoint/device_state_busy_at = 1
endpoint/send_diversion = yes
endpoint/rtp_symmetric=yes
aor/qualify_frequency = 300
aor/authenticate_qualify = no
aor/max_contacts = 3
aor/remove_existing = yes
aor/minimum_expiration = 3600
aor/support_path = yes
;phoneprov/PROFILE = profile1
endpoint/permit=0.0.0.0/0

[system-udp]
type=transport
protocol=udp
bind=0.0.0.0:5080
external_media_address=EXTERNALASTERISKHOSTNAME.COM
external_signaling_address=EXTERNALASTERISKHOSTNAME.COM
local_net=192.168.2.0/255.255.255.0

[system-tcp]
type=transport
protocol=tcp
bind=0.0.0.0:5085
external_media_address=EXTERNALASTERISKHOSTNAME.COM
external_signaling_address=EXTERNALASTERISKHOSTNAME.COM
local_net=192.168.2.0/255.255.255.0

jcolp · October 1, 2020, 2:56pm

If you are seeing a retransmission of the REGISTER with the same CSeq value that means our response never got to the endpoint, despite sending to the actual source IP address and port. They may be inspecting traffic or dropping it.

For a case with TCP you’d need to provide a full SIP trace so the SDP can be seen which dictates where media goes to as well as “rtp set debug on” which shows media being received/sent.

slvrsi · October 1, 2020, 3:10pm

Yes. I’m seeing the same CSeq value over and over… of 20… which aligns with the client not seeing the response.

The odd thing is that an older server setup using Asterisk 13 with chan_sip works over UDP… so its something different happening in PJSIP ?

As for TCP, it registered but I thought audio still occurs through UDP which is why the media (audio both ways) is not there…

But here is the logger on and debug on with TCP:
OpenWrtCLI> pjsip set logger on
PJSIP Logging enabled
OpenWrtCLI> rtp set debug on
RTP Debugging Enabled
<— Received SIP request (662 bytes) from TCP:CellphoneIP:31758 —>
REGISTER sip:EXTAsteriskHOSTNAME SIP/2.0
Via: SIP/2.0/TCP noClueWhatIp-FidoGateway?-25.108.253.33:47036;alias;branch=z9hG4bK.2sQc76vuz;rport
From: sip:username@EXTAsteriskHOSTNAME;tag=4kZTQKJRN
To: sip:username@EXTAsteriskHOSTNAME
CSeq: 20 REGISTER
Call-ID: X4nwgJptbR
Max-Forwards: 70
Supported: replaces, outbound, gruu
Accept: application/sdp
Accept: text/plain
Accept: application/vnd.gsma.rcs-ft-http+xml
Contact: sip:username@noClueWhatIp-FidoGateway?-25.108.253.33:47036;transport=tcp;+sip.instance=“urn:uuid:883f5ef2-690e-0058-9ebe-c04afeb8fa0b”;+org.linphone.specs=“groupchat,lime”
Expires: 3600
User-Agent: LinphoneAndroid/4.2 (Pixel 2) LinphoneSDK/4.3 (tags/4.3^0)
Content-Length: 0

<— Transmitting SIP response (482 bytes) to TCP:CellphoneIP:31758 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP noClueWhatIp-FidoGateway?-25.108.253.33:47036;rport=31758;received=CellphoneIP;branch=z9hG4bK.2sQc76vuz;alias
Call-ID: X4nwgJptbR
From: sip:username@EXTAsteriskHOSTNAME;tag=4kZTQKJRN
To: sip:username@EXTAsteriskHOSTNAME;tag=z9hG4bK.2sQc76vuz
CSeq: 20 REGISTER
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1601564742/29b63d6aeb743b1b277cc67b329dd791”,opaque=“3190c473302945a7”,algorithm=md5,qop=“auth”
Server: Asterisk PBX 16.3.0
Content-Length: 0

<— Received SIP request (944 bytes) from TCP:CellphoneIP:31758 —>
REGISTER sip:EXTAsteriskHOSTNAME SIP/2.0
Via: SIP/2.0/TCP noClueWhatIp-FidoGateway?-25.108.253.33:47036;alias;branch=z9hG4bK.Ai2hXynL6;rport
From: sip:username@EXTAsteriskHOSTNAME;tag=4kZTQKJRN
To: sip:username@EXTAsteriskHOSTNAME
CSeq: 21 REGISTER
Call-ID: X4nwgJptbR
Max-Forwards: 70
Supported: replaces, outbound, gruu
Accept: application/sdp
Accept: text/plain
Accept: application/vnd.gsma.rcs-ft-http+xml
Contact: sip:username@CellphoneIP:31758;transport=tcp;+sip.instance=“urn:uuid:883f5ef2-690e-0058-9ebe-c04afeb8fa0b”;+org.linphone.specs=“groupchat,lime”
Expires: 3600
User-Agent: LinphoneAndroid/4.2 (Pixel 2) LinphoneSDK/4.3 (tags/4.3^0)
Content-Length: 0
Authorization: Digest realm=“asterisk”, nonce=“1601564742/29b63d6aeb743b1b277cc67b329dd791”, algorithm=md5, opaque=“3190c473302945a7”, username=“username”, uri=“sip:EXTAsteriskHOSTNAME”, response=“c126d98e04672c66fc7966527cd67b88”, cnonce=“VNzvccW~xRqaMrxx”, nc=00000001, qop – Added contact ‘sip:username@CellphoneIP:31758;transport=TCP’ to AOR ‘username’ with expiration of 3600 seconds
<— Transmitting SIP response (525 bytes) to TCP:CellphoneIP:31758 —>
SIP/2.0 200 OK
Via: SIP/2.0/TCP noClueWhatIp-FidoGateway?-25.108.253.33:47036;rport=31758;received=CellphoneIP;branch=z9hG4bK.Ai2hXynL6;alias
Call-ID: X4nwgJptbR
From: sip:username@EXTAsteriskHOSTNAME;tag=4kZTQKJRN
To: sip:username@EXTAsteriskHOSTNAME;tag=z9hG4bK.Ai2hXynL6
CSeq: 21 REGISTER
Date: Thu, 01 Oct 2020 15:05:42 GMT
Contact: sip:username@192.168.2.200:5080;expires=3218
Contact: sip:username@CellphoneIP:31758;transport=TCP;expires=3599
Expires: 3600
Supported: path
Server: Asterisk PBX 16.3.0
Content-Length: 0

<— Received SIP request (1203 bytes) from TCP:CellphoneIP:31758 —>
INVITE sip:600@EXTAsteriskHOSTNAME SIP/2.0
Via: SIP/2.0/TCP noClueWhatIp-FidoGateway?-25.108.253.33:47036;branch=z9hG4bK.jkMYjNlQF;rport
From: sip:username@EXTAsteriskHOSTNAME;tag=1XuO7-IxJ
To: sip:600@EXTAsteriskHOSTNAME
CSeq: 20 INVITE
Call-ID: ddm2xlENeV
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 523
Contact: sip:username@CellphoneIP:31758;transport=tcp;expires=3599;+sip.instance=“urn:uuid:883f5ef2-690e-0058-9ebe-c04afeb8fa0b”;+org.linphone.specs=“groupchat,lime”
User-Agent: LinphoneAndroid/4.2 (Pixel 2) LinphoneSDK/4.3 (tags/4.3^0)

v=0
o=username 259 866 IN IP4 noClueWhatIp-FidoGateway?-25.108.253.33
s=Talk
c=IN IP4 noClueWhatIp-FidoGateway?-25.108.253.33
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 18 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 us<— Transmitting SIP response (468 bytes) to TCP:CellphoneIP:31758 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP noClueWhatIp-FidoGateway?-25.108.253.33:47036;rport=31758;received=CellphoneIP;branch=z9hG4bK.jkMYjNlQF
Call-ID: ddm2xlENeV
From: sip:username@EXTAsteriskHOSTNAME;tag=1XuO7-IxJ
To: sip:600@EXTAsteriskHOSTNAME;tag=z9hG4bK.jkMYjNlQF
CSeq: 20 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1601564748/f5520fbdbb8be6e43923f79d49c77d12”,opaque=“0303c018775c71de”,algorithm=md5,qop=“auth”
Server: Asterisk PBX 16.3.0
Content-Length: 0

<— Received SIP request (465 bytes) from TCP:CellphoneIP:31758 —>
ACK sip:600@EXTAsteriskHOSTNAME SIP/2.0
Via: SIP/2.0/TCP noClueWhatIp-FidoGateway?-25.108.253.33:47036;branch=z9hG4bK.jkMYjNlQF;rport
Call-ID: ddm2xlENeV
From: sip:username@EXTAsteriskHOSTNAME;tag=1XuO7-IxJ
To: sip:600@EXTAsteriskHOSTNAME;tag=z9hG4bK.jkMYjNlQF
Contact: sip:username@CellphoneIP:31758;transport=tcp;expires=3599;+sip.instance=“urn:uuid:883f5ef2-690e-0058-9ebe-c04afeb8fa0b”;+org.linphone.specs=“groupchat,lime”
Max-Forwards: 70
CSeq: 20 ACK
Content-Length: 0

<— Received SIP request (1490 bytes) from TCP:CellphoneIP:31758 —>
INVITE sip:600@EXTAsteriskHOSTNAME SIP/2.0
Via: SIP/2.0/TCP noClueWhatIp-FidoGateway?-25.108.253.33:47036;branch=z9hG4bK.thP-b4P-4;rport
From: sip:username@EXTAsteriskHOSTNAME;tag=1XuO7-IxJ
To: sip:600@EXTAsteriskHOSTNAME
CSeq: 21 INVITE
Call-ID: ddm2xlENeV
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 523
Contact: sip:username@CellphoneIP:31758;transport=tcp;expires=3599;+sip.instance=“urn:uuid:883f5ef2-690e-0058-9ebe-c04afeb8fa0b”;+org.linphone.specs=“groupchat,lime”
User-Agent: LinphoneAndroid/4.2 (Pixel 2) LinphoneSDK/4.3 (tags/4.3^0)
Authorization: Digest realm=“asterisk”, nonce=“1601564748/f5520fbdbb8be6e43923f79d49c77d12”, algorithm=md5, opaque=“0303c018775c71de”, username=“username”, uri=“sip:600@EXTAsteriskHOSTNAME”, response=“c36a0f878f0bb8894b5d7d3e940d5174”, cnonce="C3eEY5zXvrlPe == Setting global variable ‘SIPDOMAIN’ to ‘EXTAsteriskHOSTNAME’
<— Transmitting SIP response (294 bytes) to TCP:CellphoneIP:31758 —>
SIP/2.0 100 Trying
Via: SIP/2.0/TCP noClueWhatIp-FidoGateway?-25.108.253.33:47036;rport=31758;received=CellphoneIP;branch=z9hG4bK.thP-b4P-4
Call-ID: ddm2xlENeV
From: sip:username@EXTAsteriskHOSTNAME;tag=1XuO7-IxJ
To: sip:600@EXTAsteriskHOSTNAME
CSeq: 21 INVITE
Server: Asterisk PBX 16.3.0
Content-Length: 0

-- Executing [600@context-1282:1] Set("PJSIP/username-00000002", "userLine="*****"") in new stack
-- Executing [600@context-1282:2] Goto("PJSIP/username-00000002", "context-freePhoneLine,600,1") in new stack
-- Goto (context-freePhoneLine,600,1)
-- Executing [600@context-freePhoneLine:1] Answer("PJSIP/username-00000002", "") in new stack

<— Transmitting SIP response (792 bytes) to TCP:CellphoneIP:31758 —>
SIP/2.0 200 OK
Via: SIP/2.0/TCP noClueWhatIp-FidoGateway?-25.108.253.33:47036;rport=31758;received=CellphoneIP;branch=z9hG4bK.thP-b4P-4
Call-ID: ddm2xlENeV
From: sip:username@EXTAsteriskHOSTNAME;tag=1XuO7-IxJ
To: sip:600@EXTAsteriskHOSTNAME;tag=6cbbb678-8993-4f0a-a5da-1f9d1942d6ce
CSeq: 21 INVITE
Server: Asterisk PBX 16.3.0
Contact: sip:192.168.2.1:5085;transport=TCP
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 221